from 'keypad_entry' to 'region'. Fix the example file accordingly.
Also make some fixes in the code do reset entries on reload of the keypad.
The recently committed kpad2.jpg has the correct names.
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the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending
on the amount of time passed. The purpose is to allow the call to open up to more (or maybe
just different) members without the caller's losing his place in the queue. See
configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample
for how to associate a rule with a queue.
Along with the functional changes, new CLI and manager commands exist to show the rules defined and
there is an additional CLI command to reload the queue rules.
Future enhancements that may be made: support for realtime queue rules and support for dynamically adding
a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write
this myself very soon).
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by adding an exclamation mark to the dial string.
This patch also exists for 1.4 in the fixtoheader-1.4 branch
and has been in production for quite some time.
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- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
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This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify.
The mwimonitor option enables MWI monitoring. When the MWI state on a line changes,
then the script specified by mwimonitornotify will be executed for custom handling
of the state change, similar to the externnotify option of voicemail.conf.
Also, when the MWI state on an FXO line changes, an internal Asterisk event is
generated to indicate the new state of the associated mailbox. That may, any
module that cares about MWI information will get notified and can handle it
just as if app_voicemail had sent this notification.
(BE-253, original patch from markster, with some minor modifications by me to
add comments, documentation, and internal event support)
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This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
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1. When moh is started, we search first in memory to find the class. If we do not
find it in memory, we search realtime instead.
2. When moh is restarted (as in, it had been started on this particular channel, stopped,
and now we're starting it again), if using the "files" mode, then realtime will always
be rechecked. If you are using other modes, however, we will simply reattach to the external
running process which was playing moh earlier in the call. This is a necessary compromise so that
we don't end up with too many background processes.
3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes,
then moh classes found in realtime will be added to the in-memory list. This has the advantage
of not requiring database lookups each time moh is started, but it has the disadvantage of not
truly being realtime.
I have tested this for functionality, and it passes. I also tested this under valgrind and there
are no memory problems reported under typical use.
Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker!
(closes issue #11196, reported and patched by sergee)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line
closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
........
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines
We previously attempted to use the ESCAPE clause to set the escape delimiter to
a backslash. Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.
So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter. If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.
Reported by: elguero
Patch by: tilghman
(Closes issue #11364)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and we now have the groupcount system to implement call-limits in the dialplan. You
can use the "setvar" option in realtime/sip.conf to set limits per device.
- Implement "callcounter" as a new option to enable the call counting we need to
report device status to queue, manager and SIP subscriptions.
The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.
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way of handling DTMF in SIP. Totally undocumented, but implemented
in enough devices so we have to support it.
Code by sergee, small changes by oej.
Closes issue #11049
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line
if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line
aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It was mistakenly deleted in 1.4 without ever being merged to trunk.
Reported by eliel on #asterisk-dev.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones. The following models have been confirmed
to work: i2002, i2004 and i2050.
(closes issue #8864)
Reported by: c_hans
Patches:
chan_unistim.patch uploaded by c (license 304)
ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
phones to be rung in a specific order.
(closes issue #7279, reported and initially patched by diLLec, patch reworked by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ruffle
Patches:
app_voicemail.c.diff uploaded by ruffle (license 201)
10739-moveheard.diff uploaded by qwell (license 4)
Tested by: callguy, ruffle
Add an option to disable the automatic moving of "heard" messages to the Old folder.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: snar
Patches:
app-queue-cdr-trunk.patch uploaded by snar (license 245)
queues.conf.patch uploaded by snar (license 245)
Add an updatecdr option to queues.conf, so that if a "member name" is specified,
the cdr record will be updated with that, rather than the channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: junky
Patches:
register_trying.diff.txt uploaded by jcmoore
Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
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This feature may be turned on by adding imapgreetings=yes to the general section of voicemail.conf
voicemail.conf.sample has details on the options added.
As a result, IMAP storage now has RETRIEVE and DISPOSE macros defined.
In addition to the IMAP greeting changes, I also have added an enum for the voicemail folders
and so now the code should be easier to understand and maintain when it comes to this area.
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Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
This permission was discussed on the -dev mailing list some months back.
Issue 8613, patch by johann8384, with some minor changes by me.
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disclaimer along with SIP messages in the header, X-Disclaimer. This is off by
default. Also, the text of the disclaimer can be customized in sip.conf.
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features.conf. This allows you to create a feature one time, and then map it
into groups for various different key mappings for the same feature, as well
as easy access control to groups of features.
(patch from bbryant)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tilghman Lesher
Luigi Rizzo
Mark Michelson
Russell Bryant
Olle Johansson
Jason Parker
Joshua Colp
Kevin P. Fleming
Steve Murphy
Christian Richter
Matthew Fredrickson
Philippe Sultan
Dwayne M. Hubbard